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1. 发明授权

EP2625688B1 APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL AND FOR PROVIDING A HIGHER TEMPORAL GRANULARITY FOR A COMBINED UNIFIED SPEECH AND AUDIO CODEC (USAC) 有权转让
标题翻译：用于处理信号和音频设备和方法，以提供更高的分辨率时间对于组合的单一的语音和音频编解码（USAC）
公开(公告)号：EP2625688B1
公开(公告)日：2014-12-03
申请号：EP11764739.6
申请日：2011-10-04
申请人： Fraunhofer Gesellschaft zur Förderung der angewandten Forschung e.V. , VoiceAge Corporation
发明人： MULTRUS, Markus , GRILL, Bernhard , NEUENDORF, Max , RETTELBACH, Nikolaus , FUCHS, Guillaume , GOURNAY, Philippe , LEFEBVRE, Roch , BESSETTE, Bruno , WILDE, Stephan
IPC分类号： G10L21/00 , G10L19/12 , G10L19/02
CPC分类号： G10L19/12 , G10L19/0204 , G10L21/00 , G10L2019/0012

2. 发明授权

EP1688920B1 Speech signal decoding 有权
标题翻译：语音信号解码
公开(公告)号：EP1688920B1
公开(公告)日：2010-04-07
申请号：EP06112489.7
申请日：2000-10-31
申请人： NEC CORPORATION
发明人： Murashima, Atsushi
IPC分类号： G10L21/02 , G10L19/14
CPC分类号： G10L19/083 , G10L21/0364 , G10L2019/0012
摘要： The quality of reconstructed speech on which background noise is superimposed is improved in a speech signal decoding apparatus for generating a speech signal by driving a filter, which is constituted by linear prediction coefficients, by an excitation signal. A smoothing circuit smoothes sound source gain in a noise segment using sound source gain that was obtained in the past. A smoothing-quantity limiting circuit calculates an amount of fluctuation represented by dividing, by the sound source gain, the absolute value of the difference between the sound source gain and the sound source gain that has been smoothed, and limits the value of the smoothed gain in such a manner that the amount of fluctuation will not exceed a certain threshold value.

3. 发明授权

EP0450064B2 DIGITAL SPEECH CODER HAVING IMPROVED SUB-SAMPLE RESOLUTION LONG-TERM PREDICTOR 失效
标题翻译：具有改进的长期预测BY子样本NUMERIC讲话
公开(公告)号：EP0450064B2
公开(公告)日：2006-08-09
申请号：EP91905041.9
申请日：1990-06-25
申请人： MOTOROLA, INC.
发明人： GERSON, Ira, Alan , JASIUK, Mark, A.
IPC分类号： G10L19/12
CPC分类号： G10L19/12 , G10L2019/0011 , G10L2019/0012
摘要： A digital speech coder includes a long-term filter (124) having an improved sub-sample resolution long-term predictor which allows for subsample resolution for the lag parameter L. A frame of N samples of input speech vector s(n) is applied to an adder (510). The output of the adder (510) produces the output vector b(n) for the long term filter (124). The output vector b(n) is fed back to a delayed vector generator block (530) of the long-term predictor. The nominal long-term predictor lag parameter L is also input to the delayed vector generator block (530). The long-term predictor lag parameter L can take on non-integer values, which may be multiples of one half, one third, one fourth or any other rational fraction. The delayed vector generator (530) includes a memory which holds past samples of b(n). In addition, interpolated samples of b(n) are also calculated by the delayed vector generator (530) and stored in its memory, at least one interpolated sample being calculated and stored between each past sample of b(n). The delayed vector generator (530) provides output vector q(n) to the long-term multiplier block (520), which scales the long-term predictor response by the long-term predictor coefficient beta . The scaled output beta q(n) is then applied to the adder (510) to complete the feedback loop of the recursive filter (124).

4. 发明公开

EP1596367A2 Method and apparatus for speech decoding 审中-公开
标题翻译： Verfahren und Vorrichtung zur Dekodierung von Sprachsignalen
公开(公告)号：EP1596367A2
公开(公告)日：2005-11-16
申请号：EP05015792.4
申请日：1998-12-07
申请人： MITSUBISHI DENKI KABUSHIKI KAISHA
发明人： Yamaura, Tadashi, Mitsubishi Denki K.K.
IPC分类号： G10L19/12
CPC分类号： G10L19/135 , G10L13/02 , G10L19/012 , G10L19/06 , G10L19/083 , G10L19/09 , G10L19/107 , G10L19/12 , G10L19/125 , G10L19/18 , G10L21/0264 , G10L25/93 , G10L2019/0002 , G10L2019/0005 , G10L2019/0007 , G10L2019/0011 , G10L2019/0012 , G10L2019/0016
摘要： A speech decoding method according to a code-excited linear prediction (CELP) speech.
Obtaining a time series vector from an excitation codebook and determining according to a decoded gain if a modification of said vector is necessary. If required the vector is modified such that its number of zero - amplitude valued samples is changed.
摘要翻译：根据码激励线性预测（CELP）语音的语音解码方法。从激励码本获取时间序列向量，并且如果需要修改所述向量，则根据解码增益来确定。如果需要，该矢量被修改为使其零幅度值样本的数量改变。

5. 发明公开

EP1480201A2 Reduction of quantization-induced block-discontinuities in an audio coder 有权
标题翻译： EINEM AUDIO-KODIERER中的VERRINGERUNG DER DATENBLOCK-UNTERBRECHUNGEN VON QUANTISIERUNG
公开(公告)号：EP1480201A2
公开(公告)日：2004-11-24
申请号：EP04076676.8
申请日：2000-05-25
申请人： America Online, Inc.
发明人： Wu, Shuwu , Mantegna, John , Perlmutter, Keren
IPC分类号： G10L19/02
CPC分类号： G10L19/00 , G10L19/0212 , G10L19/022 , G10L19/028 , G10L19/038 , G10L2019/0012
摘要： A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block-discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters form the noise and weak signal components (collectively called residue); and adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm. The invention further includes corresponding computer program implementations of these and other algorithms.
摘要翻译：一种用于减少由连续信号，特别是音频信号的有损压缩和解压缩引起的量化引起的块不连续性的方法和系统。一个实施例包括通用的，超低延迟的高效音频编解码算法。更具体地说，本发明包括一种使用新颖的边界分析和合成框架来压缩和解压缩音频信号的方法和装置，以大幅度减少量化引起的帧或块不连续性; 一种新颖的自适应余弦分组变换（ACPT）作为选择的有效捕获输入音频特性的变换; 用于分离强信号簇的信号残差分类器形成噪声和弱信号分量（统称为残差）; 和用于信号分量的自适应稀疏矢量量化（ASVQ）算法; 残留物的随机噪声模型; 和相关联的速率控制算法。本发明还包括这些和其他算法的相应的计算机程序实现。

6. 发明公开

EP1204092A3 Speech decoder capable of decoding background noise signal with high quality 有权
标题翻译：语音解码器与背景噪声的重放
公开(公告)号：EP1204092A3
公开(公告)日：2003-11-19
申请号：EP01125496.8
申请日：2001-11-06
申请人： NEC CORPORATION
发明人： Ozawa, Kazunori
IPC分类号： G10L19/00 , G10L19/14
CPC分类号： G10L19/083 , G10L19/06 , G10L2019/0012
摘要： In response to a coded speech signal output from a speech coder, a speech decoder decodes the coded speech signal into a reproduction speech signal. If the reproduction speech signal meets predetermined conditions, for example, "silence", "unvoiced sound", and the like, the speech decoder further operates as the following. The speech decoder calculates spectral parameters based on the reproduction speech signal, and calculates an excitation signal on the basis of the reproduction speech signal and the spectral parameters. In the calculation, a level of the excitation signal is also obtained. The speech decoder smoothes in time at least one of the spectral parameters and the level of the excitation signal. The speech decoder synthesizes the excitation signal by using the synthesis filter constructed with the spectrum parameters, so as to reproduce the speech signal. The speech signal has an excellent quality even if a bit rate is low.

7. 发明授权

EP0793218B1 Speech synthesis method and apparatus 失效
标题翻译：语音合成方法和装置
公开(公告)号：EP0793218B1
公开(公告)日：2003-04-23
申请号：EP97301003.6
申请日：1997-02-17
申请人： SONY CORPORATION
发明人： Inoue, Akira , Nishiguchi, Masayuki
IPC分类号： G10L13/00
CPC分类号： G10L19/07 , G10L19/06 , G10L2019/0012
摘要： A speech synthesis apparatus in which spectrum emphasis characteristics can be set easily taking into account the frequency response and psychoacoustic hearing sense and in which the degree of freedom in setting the response is larger. An excitation signal ex(n) is synthesized by a synthesis filter 12 to give a synthesized speech signal which is sent to a spectrum emphasis filter 13. The spectrum emphasis filter 13 spectrum-emphasizes the synthesized speech signal and outputs the resulting spectrum-emphasized signal. The vocal tract parameters from an input terminal 21 are converted by a parameter conversion circuit 23 into linear spectral pair (LSP) frequencies which are interpolated by an LSP interpolation circuit 24 with equal-interval line spectral pair frequencies to produce interpolated LSP frequencies. The transfer function of the spectrum emphasis filter 13 is determined on the basis of the interpolated LSP frequencies.

8. 发明公开

EP1235203A3 Method for concealing erased speech frames and decoder therefor 有权
标题翻译：在语音帧和解码器Dafer的损失的情况下隐藏方法
公开(公告)号：EP1235203A3
公开(公告)日：2002-09-11
申请号：EP02100190.4
申请日：2002-02-26
申请人： Texas Instruments Incorporated
发明人： UNNO, Takahiro
IPC分类号： G10L19/00 , G10L19/12
CPC分类号： G10L19/005 , G10L19/12 , G10L2019/0011 , G10L2019/0012 , G10L2025/935
摘要： A decoder for code excited LP encoded frames with both adaptive and fixed codebooks; erased frame concealment uses repetitive excitation plus a smoothing of pitch gain in the next good frame, plus multilevel voicing classification with multiple thresholds of correlations determining linear interpolated adaptive and fixed codebook excitation contributions.

9. 发明公开

EP1204092A2 Speech decoder capable of decoding background noise signal with high quality 有权
标题翻译： Sprachdekoder zum hochqualitativen Dekodieren von Signalen mit Hintergrundrauschen
公开(公告)号：EP1204092A2
公开(公告)日：2002-05-08
申请号：EP01125496.8
申请日：2001-11-06
申请人： NEC CORPORATION
发明人： Ozawa, Kazunori
IPC分类号： G10L19/00
CPC分类号： G10L19/083 , G10L19/06 , G10L2019/0012
摘要： In response to a coded speech signal output from a speech coder, a speech decoder decodes the coded speech signal into a reproduction speech signal. If the reproduction speech signal meets predetermined conditions, for example, "silence", "unvoiced sound", and the like, the speech decoder further operates as the following. The speech decoder calculates spectral parameters based on the reproduction speech signal, and calculates an excitation signal on the basis of the reproduction speech signal and the spectral parameters. In the calculation, a level of the excitation signal is also obtained. The speech decoder smoothes in time at least one of the spectral parameters and the level of the excitation signal. The speech decoder synthesizes the excitation signal by using the synthesis filter constructed with the spectrum parameters, so as to reproduce the speech signal. The speech signal has an excellent quality even if a bit rate is low.
摘要翻译：响应于从语音编码器输出的编码语音信号，语音解码器将编码的语音信号解码为再现语音信号。如果再现语音信号满足预定条件，例如“静音”，“无声音”等，语音解码器进一步如下操作。语音解码器基于再现语音信号来计算频谱参数，并且基于再现语音信号和频谱参数来计算激励信号。在计算中，也获得激励信号的电平。语音解码器在时间上平滑频谱参数和激励信号的电平中的至少一个。语音解码器通过使用由频谱参数构成的合成滤波器来合成激励信号，以再现语音信号。即使比特率低，语音信号也具有优良的品质。

10. 发明授权

EP0673018B1 Linear prediction coefficient generation during frame erasure or packet loss 失效
标题翻译：在数据帧中的失败或数据分组的丢失的情况下产生的线性预测系数
公开(公告)号：EP0673018B1
公开(公告)日：2001-10-04
申请号：EP95301488.3
申请日：1995-03-08
申请人： AT&T Corp.
发明人： Chen, Juin-Hwey , Watkins, Craig Robert
IPC分类号： G10L19/00 , G10L19/12
CPC分类号： G10L19/005 , G10L19/06 , G10L19/12 , G10L2019/0012
摘要： A speech coding system robust to frame erasure (or packet loss) is described. Illustrative embodiments are directed to a modified version of CCITT standard G.728. In the event of frame erasure, vectors of an excitation signal are synthesized based on previously stored excitation signal vectors generated during non-erased frames. This synthesis differs for voiced and non-voiced speech. During erased frames, linear prediction filter coefficients are synthesized as a weighted extrapolation of a set of linear prediction filter coefficients determined during non-erased frames. The weighting factor is a number less than 1. This weighting accomplishes a bandwidth-expansion of peaks in the frequency response of a linear predictive filter. Computational complexity during erased frames is reduced through the elimination of certain computations needed during non-erased frames only. This reduction in computational complexity offsets additional computation required for excitation signal synthesis and linear prediction filter coefficient generation during erased frames.

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